[Asterisk-Users] how to get caller ID

Sat Sep 18 19:16:36 MST 2004

thanx andrew
first of all
your messages are in Plain Text format!

i hv monitored Asterisk both managerAPI console and Asterisk main console to
see wht is actually going on .when a new incoming connection comes.

when the phone is ringing.it gives

starting simple swithc on 'ZAP/1-1'

and on manager API i get newExten event with exten 's'
from channel 'zap/1-1'

i hvnt picked up my fone.

i think its its ringing 7-8 times and asterisk doesnt do anything .

----- Original Message -----
From: <asterisk-users-request at lists.digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, September 16, 2004 9:22 AM
Subject: Asterisk-Users Digest, Vol 2, Issue 152

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> Today's Topics:
>
>    1. Re: E3 PCI Cards (Noah Miller)
>    2. spandsp on current cvs? (Rich Adamson)
>    3. Re: Broadvoice BYOD Plans - 3-way and Call Waiting (Chris Shaw)
>    4. Re: No Caller Name sent from Asterisk over National or
>       DMS100? (Jason Kawakami)
>    5. RE: Intertex IX66 (Chris HARIGA)
>    6. how to get caller ID  (vrushank)
>    7. Re: how to get caller ID (Andrew Thompson)
>    8. Re: E3 PCI Cards (Benjamin on Asterisk Mailing Lists)
>    9. Beyond T1 (Christopher Jacob)
>   10. call parking & forwarding (Maros RAJNOCH)
>   11. What can you do with Asterisk in Brazil following the law
>       (Johannes van Hulst)
>   12. ID for outgoing calls from DDI (DID) line (Maros RAJNOCH)
>   13. Re: Beyond T1 (Andrew Thompson)
>   14. Re: Beyond T1 (Steven P. Donegan)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 16 Sep 2004 11:03:48 -0400
> From: Noah Miller <noah at rosecompanies.com>
> Subject: Re: [Asterisk-Users] E3 PCI Cards
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <9D1CF220-07F1-11D9-9D68-000393971C6E at rosecompanies.com>
> Content-Type: text/plain; charset=US-ASCII; format=flowed
>
> >> Another promising candidate is Apple's dual G5 (PPC970) Xserve (a 1U
> >> server).
> >> http://www.apple.com/xserve
> >> this one looks as if it might beat the price/performance ratio of a
> >> high end Intel server.
> >
> > The Apple G5 Xserv system has a "PCI-X" interface.  Does anyone know
> > what that is and will a T405P or T410P card work?
> >
> >> Both systems run LinuxPPC.
> >
> > Does anyone have * running on PPC?
>
> Yeah, check out:
>
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20MacOSX%20Support
>
> Specifically for OS X.  There's a download link.  The problem still is
> that no one has written ppc drivers for the Digium cards.  As I
> understand, the only drivers are for GNU/Linux on i386.  You wanna
> write some for the good of the BSD and PPC communities? ;-)
>
>
>
>
>
> >
> >
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> ------------------------------
>
> Message: 2
> Date: Thu, 16 Sep 2004 10:06:39 -0600
> From: Rich Adamson <radamson at routers.com>
> Subject: [Asterisk-Users] spandsp on current cvs?
> To: Asterisk-a-users-list <asterisk-users at lists.digium.com>
> Message-ID: <Chameleon.1095347315.adar0 at vegas>
> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1
>
>
> Steve or anyone...
>
> Will spandsp install on the current cvs?
>
> Looked like the code at ftp.opencall.org/pub/spandsp was intended
> to be applied to the old stable release. Anyone know?
>
> Rich
>
>
>
>
> ------------------------------
>
> Message: 3
> Date: Thu, 16 Sep 2004 08:11:10 -0700
> From: "Chris Shaw" <chriss at watertech.com>
> Subject: Re: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call
> Waiting
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <0d5d01c49bff$661e8b20$8805640a at chriss>
> Content-Type: text/plain; charset="iso-8859-1"
>
> > The other issue is that call waiting does not appear to work. The way
I'm
> > expecting it to work with Asterisk is to send the second call to me -
I'm
> > using SetGroup and CheckGroup within Asterisk to limit my calls to two
at
> a
> > time total. However, if I'm on a phone call (incoming or outgoing),
> Broadvoice
> > transfers a second call to a "person you are calling is busy" message --
I
> > don't see any additional SIP traffic to the Asterisk box.
>
> You must have call waiting turned off on your comm pilot control panel, go
> to www.broadvoice.com and log into your control panel and make sure call
> waiting is turned on.
>
>     -Chris
>
>
>
> ------------------------------
>
> Message: 4
> Date: Thu, 16 Sep 2004 09:20:09 -0600
> From: "Jason Kawakami" <jkkawakami at optellabs.com>
> Subject: [Asterisk-Users] Re: No Caller Name sent from Asterisk over
> National or DMS100?
> To: <asterisk-users at lists.digium.com>
> Message-ID: <001501c49c00$ab8cc900$2201a8c0 at jkk00298l>
> Content-Type: text/plain; charset="iso-8859-1"
>
>
> ----- Original Message -----
> > Message: 3
> > Date: Thu, 16 Sep 2004 07:57:15 -0400 (EDT)
> > From: David Troy <dave at popvox.com>
> > Subject: Re: [Asterisk-Users] No Caller Name sent from Asterisk over
> > National or DMS100 PRI to a Norstar MICS?
> > snip>
> > > I have a PRI link up and running between Asterisk and a Nortel Norstar
> MICS
> > > v4.1 . I'm having a problem getting the textual Caller Name across the
> link
> > > from Ast to Ns, however numeric Caller ID arrives and displays fine.
> >From Ns
> > > to Ast both elements come through fine. I'm forcing dummy values for
> testing
> > > using:
> <snip>
>
> everyone remember that we are talking about a private connection here.  if
i
> read the original post here correctly the issue is between the * and the
> Norstar not out to the PSTN.
>
> i have been tying NEC's together for 15+ years with a proprietary ISDN
> protocol that sends station name across the d-channel without any reverse
> lookup DB.
>
> Now that being said I am no expert on d-channel messaging so I can't
really
> answer the question on how/if we can pass the CALLERIDNAME across a
private
> d-channel connection between * and another PBX.
>
> Jason Kawakami
> www.optellabs.com
>
>
>
> ------------------------------
>
> Message: 5
> Date: Thu, 16 Sep 2004 11:21:41 -0400
> From: "Chris HARIGA" <contact at techselesta.com>
> Subject: RE: [Asterisk-Users] Intertex IX66
> To: "'Jason Williams'" <jas.williams at gmail.com>, "'Asterisk Users
> Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID:
>
<!~!UENERkVCMDkAAQACAAAAAAAAAAAAAAAAABgAAAAAAAAAgvaz9VLBY0Wot+jOKFJUmMKAAAAQ
AAAAPMRrmtinMkSwRXqda6FZHQEAAAAA at techselesta.com>
>
> Content-Type: text/plain; charset="us-ascii"
>
> Lolllllll,
>
> That's a good one :))
>
> U make my day :)
>
> Best regards,
>
> Chris HARIGA
>
> P.S.: I send my ethereal log to Intertex.se and I hope to fix the problem
> asap. I will post on the list the "solution".
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jason
Williams
> Sent: Thursday, September 16, 2004 4:50 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Intertex IX66
>
> On Thu, 9 Sep 2004 12:26:56 -0400, Chris HARIGA <contact at techselesta.com>
> wrote:
> > Hi,
> >
> > I have Asterisk w/ 192.168.1.1 and I setup IX66 to be 192.168.1.2 (I'm
> using
> > pppoe client and dyndns.org on IX66)
> > I setup on Local DNS Server my * box and after that I was able to
register
> > my phones from the Internet.
> > I cannot understand my problem with one way sound... what is wrong on my
> > configuration :((
>
> As the IX66 is a sip aware router make sure you have no entries for
> nat in your sip.conf, and let the ix66 deal with the nat, not * . I
> hope this helps.
>
>
> Jason
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
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> ------------------------------
>
> Message: 6
> Date: Fri, 17 Sep 2004 21:05:08 -0700
> From: "vrushank" <vrushank at varaha.com>
> Subject: [Asterisk-Users] how to get caller ID
> To: <asterisk-users at lists.digium.com>
> Message-ID: <001001c49d34$afbf0c10$7a00a8c0 at VRUSHANK>
> Content-Type: text/plain; charset="iso-8859-1"
>
> i cannot see caller ID of the call originated from outside zap channel.
> i hv configured both zapata.conf and extensions.conf.
> i m right now in india
> i think asterisk only supports Bellcore enable caller ID.
> so is it  the same bug of BT caller ID problem in UK?
> or it is the bug of my asterisk configuration?
> i hv enabled callerID from my TELCO.
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> ------------------------------
>
> Message: 7
> Date: Thu, 16 Sep 2004 11:45:21 -0400
> From: Andrew Thompson <asteriskuser at aktzero.com>
> Subject: Re: [Asterisk-Users] how to get caller ID
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4149B511.9080001 at aktzero.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> vrushank wrote:
> > i cannot see caller ID of the call originated from outside zap channel.
> > i hv configured both zapata.conf and extensions.conf.
> > i m right now in india
> > i think asterisk only supports Bellcore enable caller ID.
> > so is it  the same bug of BT caller ID problem in UK?
> > or it is the bug of my asterisk configuration?
> > i hv enabled callerID from my TELCO.
>
> Have you monitored the console while the line is ringing to verify that
> asterisk is not seeing the callerid and not paying attention to it?
>
>
> PS: I'm testing a new email client, please forgive me if this message is
> not in Plain Text. (And someone please let me know!)
>
> --
> Andrew Thompson
> http://aktzero.com/
>
>
> ------------------------------
>
> Message: 8
> Date: Fri, 17 Sep 2004 00:45:55 +0900
> From: Benjamin on Asterisk Mailing Lists
> <benjk.on.asterisk.ml at gmail.com>
> Subject: Re: [Asterisk-Users] E3 PCI Cards
> To: Noah Miller <noah at rosecompanies.com>
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <10913b9f040916084562647110 at mail.gmail.com>
> Content-Type: text/plain; charset=US-ASCII
>
> On Thu, 16 Sep 2004 11:03:48 -0400, Noah Miller <noah at rosecompanies.com>
wrote:
> > > Does anyone have * running on PPC?
> >
> > http://www.voip-info.org/tiki-index.php?page=Asterisk%20MacOSX%20Support
> >
> > Specifically for OS X.  There's a download link.  The problem still is
> > that no one has written ppc drivers for the Digium cards.  As I
> > understand, the only drivers are for GNU/Linux on i386.
>
> That's not entirely correct. The Zaptel drivers work on LinuxPPC.
>
> Further, there is some work in progress on Zaptel drivers for BSD and
> some folks use X100P and TDM400 on FreeBSD already. Since OSX is BSD
> based, it will eventually benefit from the work done to bring Zaptel
> to BSD. We have made an Xserve available for Rich Murphey, one of the
> main contributors to the Asterisk on BSD effort, specifically for him
> to test things on OSX.
>
> What's needed is more contributors to the BSD effort, or so it would
> seem. Since driver development requires skills that are less common
> than those required for many other development tasks, there are fewer
> people who can do it. It also takes more time to move drivers from one
> platform to another. I think a sponsorship fund could do some good
> because it might give somebody the ability to work fulltime on drivers
> for BSD in general and OSX in particular.
>
> I believe that it should be possible to raise significant sponsorship
> funds for drivers (especially for OSX) from end user donations alone.
> In order to do that, a few people need to come together, think about
> how to organise this, set up a website, open a kagi and/or paypal
> account and get the word out. I am discussing this idea at present
> with some Mac folks who seem to be willing to put a bit of time and
> effort into this. Anybody who would like to join in on this, please
> contact me directly.
>
> rgds
> benjk
>
> --
> Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
> Tokyo, Japan.
>
> NB: Spam filters in place. Messages unrelated to the * mailing lists
> may get trashed.
>
>
> ------------------------------
>
> Message: 9
> Date: Thu, 16 Sep 2004 11:54:38 -0400
> From: "Christopher Jacob" <chris at jacob-solutions.com>
> Subject: [Asterisk-Users] Beyond T1
> To: <asterisk-users at lists.digium.com>
> Message-ID: <20040916155435.9215A2FD3FD at lists.digium.com>
> Content-Type: text/plain; charset="us-ascii"
>
> All,
>
> This may be a stupid question, but here it is...
>
> What interface gives the most density? Do I top out at T1's? For instance,
4
> t1's to the Digium Quad span t1 card. Is there an interface available for
T3
> or DS3?
>
> Thanks,
>
> Chris
>
>
>
>
>
>
> ------------------------------
>
> Message: 10
> Date: Thu, 16 Sep 2004 18:02:51 +0200
> From: Maros RAJNOCH <ml_asterisk-users at rajnoch.sk>
> Subject: [Asterisk-Users] call parking & forwarding
> To: ASTERISK <asterisk-users at lists.digium.com>
> Message-ID: <20040916160251.GA13901 at iqvision.sk>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi everbody,
>
> I have problem with configuring call parking and forwarding.
>
> firstly my setup:
> I have one asterisk with gnu-gatekeeper on the same PC.
> As phones I use voip-phones with H323 support.
>
> phones are registered on gatekeeper as terminal and
> asterisk as gateway.
>
>
> I setup features.conf (parking.conf) like:
>
> [general]
> parkext => 700
> parkpos => 701-720
> context => parkedcalls
> parkingtime => 45
>
> and include parkedcalls context to extensions.conf
> but without any success
>
> for example: somebody call me from PSTN, and I pick up call on my h323
phone in room #1
> Now I want to go to another room (room#2), so I dial #700 (in hope to
transfer call to parking queue)
> At this time I hear tone (one tone for any one keystroke -- I think tones
are simulated by phone - not by asterisk)
> but nothing to happen. Also no records in asterisk logs.
>
> Have anybody idea what may be wrong?
>
> Another situation: call forward.
>
> I have no idea how to do it. There's no any reference in any
documentation!?
> I mean: Somebody call me from PSTN and I pick up this call by my h323
phone.
> Now I want forward this call to my colleague to another h323 phone.
>
> ANY IDEA HOW TO DO IT?
>
> Thanks for any help.
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> ------------------------------
>
> Message: 11
> Date: Thu, 16 Sep 2004 13:06:10 -0300
> From: "Johannes van Hulst" <Han.vanHulst at Terra.com.br>
> Subject: [Asterisk-Users] What can you do with Asterisk in Brazil
> following the law
> To: <asterisk-users at lists.digium.com>
> Message-ID: <20040916160653.727EB3C013 at arica.terra.com.br>
> Content-Type: text/plain; charset="us-ascii"
>
> Has anybody any idea what I can do with asterisk following the Brazilian
> law.
>
> I do not have a multimedia license or a telecom license, but I ace
asterisk.
>
>
>
> Are there companies who are looking for asterisk expertise in Rio de
> Janeiro?
>
>
>
>
>
> Greeting Han
>
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> ------------------------------
>
> Message: 12
> Date: Thu, 16 Sep 2004 18:16:46 +0200
> From: Maros RAJNOCH <ml_asterisk-users at rajnoch.sk>
> Subject: [Asterisk-Users] ID for outgoing calls from DDI (DID) line
> To: ASTERISK <asterisk-users at lists.digium.com>
> Message-ID: <20040916161646.GB13901 at iqvision.sk>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi again,
>
> in my * I have one ISDN BRI line with DID (DDI) preselection.
> so in fact I have 100 extensions: +421 33 12 34 56 xx
>
> Q: Is in my power -- or in power of * -- to influence which of these
> extensions will occur in my cellular display?
>
> THANKS.
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> ------------------------------
>
> Message: 13
> Date: Thu, 16 Sep 2004 12:17:20 -0400
> From: Andrew Thompson <asteriskuser at aktzero.com>
> Subject: Re: [Asterisk-Users] Beyond T1
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4149BC90.1040509 at aktzero.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Christopher Jacob wrote:
> > All,
> >
> > This may be a stupid question, but here it is...
> >
> > What interface gives the most density? Do I top out at T1's? For
instance, 4
> > t1's to the Digium Quad span t1 card. Is there an interface available
for T3
> > or DS3?
>
> Depending on where you using the circuits, you might try an E1. It uses
> the same total bandwidth as a T1(I think), but splits the channels at
> 56K instead of 64K, yielding more channels. (And now I can't remember
> the number.)
>
> I haven't heard of direct DS3 connectivity...
>
> Just stretching my imagination a little bit, you might be able to plug a
>   DS3 into a H323 box, and then feed the IP-side of the calls to
> asterisk....
>
> --
> Andrew Thompson
> http://aktzero.com/
>
>
> ------------------------------
>
> Message: 14
> Date: Thu, 16 Sep 2004 09:19:57 -0700
> From: "Steven P. Donegan" <steve at donegan.org>
> Subject: Re: [Asterisk-Users] Beyond T1
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4149BD2D.40902 at donegan.org>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Andrew Thompson wrote:
>
> > Christopher Jacob wrote:
> >
> >> All,
> >>
> >> This may be a stupid question, but here it is...
> >>
> >> What interface gives the most density? Do I top out at T1's? For
> >> instance, 4
> >> t1's to the Digium Quad span t1 card. Is there an interface available
> >> for T3
> >> or DS3?
> >
> >
> > Depending on where you using the circuits, you might try an E1. It
> > uses the same total bandwidth as a T1(I think), but splits the
> > channels at 56K instead of 64K, yielding more channels. (And now I
> > can't remember the number.)
> >
> > I haven't heard of direct DS3 connectivity...
> >
> > Just stretching my imagination a little bit, you might be able to plug
> > a  DS3 into a H323 box, and then feed the IP-side of the calls to
> > asterisk....
> >
> Actually T1 is 24x64k and E1 is 30x64k - 1.536 megabits/sec -vs- 2.0 if
> I recall correctly...
>
>
>
> ------------------------------
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> End of Asterisk-Users Digest, Vol 2, Issue 152
> **********************************************